[{"data":1,"prerenderedAt":152},["ShallowReactive",2],{"home":3},{"posts":4,"talks":5,"writing":88,"projects":101},[],[6,30,44,60,75],{"id":7,"title":8,"body":9,"coSpeakers":16,"description":13,"draft":17,"event":18,"eventUrl":16,"extension":19,"location":16,"meta":20,"navigation":21,"originalLanguage":22,"path":23,"role":24,"seo":25,"slides":16,"stem":26,"summaryOnly":17,"talkDate":27,"translatedAt":16,"video":28,"__hash__":29},"talks\u002Ftalks\u002Fchromium-webrtc-bitrate.md","How Chromium Manages WebRTC Bitrate",{"type":10,"value":11,"toc":12},"minimark",[],{"title":13,"searchDepth":14,"depth":14,"links":15},"",2,[],null,false,"VideoTech 2024","md",{},true,"ru","\u002Ftalks\u002Fchromium-webrtc-bitrate","speaker",{"title":8,"description":13},"talks\u002Fchromium-webrtc-bitrate","2024-09","https:\u002F\u002Fwww.youtube.com\u002Fwatch?v=4G6Cv1xN-aI","zanUUpk4AsG5xMDpldqPwl0K4iF14Sde3ZNvF0e1N3Y",{"id":31,"title":32,"body":33,"coSpeakers":16,"description":13,"draft":17,"event":18,"eventUrl":37,"extension":19,"location":16,"meta":38,"navigation":21,"originalLanguage":22,"path":39,"role":40,"seo":41,"slides":16,"stem":42,"summaryOnly":21,"talkDate":27,"translatedAt":16,"video":16,"__hash__":43},"talks\u002Ftalks\u002Fhigh-load-video-service.md","How to Prepare for High Load in a Video Service?",{"type":10,"value":34,"toc":35},[],{"title":13,"searchDepth":14,"depth":14,"links":36},[],"https:\u002F\u002Fvtconf.com\u002Fen\u002Farchive\u002F2024\u002Ftalks\u002F20006146-how-to-prepare-for-high-load-in-a-video-service\u002F",{},"\u002Ftalks\u002Fhigh-load-video-service","host",{"title":32,"description":13},"talks\u002Fhigh-load-video-service","2vORRnUYy1xk-FVlF1ZNDyAIwpXCWM4ZHLze9c3exJo",{"id":45,"title":46,"body":47,"coSpeakers":16,"description":13,"draft":17,"event":51,"eventUrl":16,"extension":19,"location":16,"meta":52,"navigation":21,"originalLanguage":22,"path":53,"role":54,"seo":55,"slides":16,"stem":56,"summaryOnly":21,"talkDate":57,"translatedAt":16,"video":58,"__hash__":59},"talks\u002Ftalks\u002Fwebrtc-quality-control-interview.md","Interview: Quality Control of WebRTC Sessions",{"type":10,"value":48,"toc":49},[],{"title":13,"searchDepth":14,"depth":14,"links":50},[],"VideoTech 2023",{},"\u002Ftalks\u002Fwebrtc-quality-control-interview","interviewee",{"title":46,"description":13},"talks\u002Fwebrtc-quality-control-interview","2023-11","https:\u002F\u002Fwww.youtube.com\u002Fwatch?v=UwTD7axZ8KM","dqd7x5GdBXffyQ3CKRN3jooVH8CqtfxOifnTYu4xOK8",{"id":61,"title":62,"body":63,"coSpeakers":16,"description":13,"draft":17,"event":67,"eventUrl":16,"extension":19,"location":16,"meta":68,"navigation":21,"originalLanguage":22,"path":69,"role":24,"seo":70,"slides":16,"stem":71,"summaryOnly":17,"talkDate":72,"translatedAt":16,"video":73,"__hash__":74},"talks\u002Ftalks\u002Fugc-video-testimonials.md","Modern UGC: Collecting Video Testimonials in the Browser without WebRTC",{"type":10,"value":64,"toc":65},[],{"title":13,"searchDepth":14,"depth":14,"links":66},[],"TechTrain 2022",{},"\u002Ftalks\u002Fugc-video-testimonials",{"title":62,"description":13},"talks\u002Fugc-video-testimonials","2022","https:\u002F\u002Fwww.youtube.com\u002Fwatch?v=ZY9AcASx2mc","H4hL88QgUDJAkb9CMWHpZmkuytRAcLqaydzJYKVO6vo",{"id":76,"title":77,"body":78,"coSpeakers":16,"description":13,"draft":17,"event":67,"eventUrl":16,"extension":19,"location":16,"meta":82,"navigation":21,"originalLanguage":22,"path":83,"role":24,"seo":84,"slides":16,"stem":85,"summaryOnly":17,"talkDate":72,"translatedAt":16,"video":86,"__hash__":87},"talks\u002Ftalks\u002Fwebrtc-statistics.md","Making Friends with WebRTC Statistics and Saving Thousands of Debugging Hours",{"type":10,"value":79,"toc":80},[],{"title":13,"searchDepth":14,"depth":14,"links":81},[],{},"\u002Ftalks\u002Fwebrtc-statistics",{"title":77,"description":13},"talks\u002Fwebrtc-statistics","https:\u002F\u002Fwww.youtube.com\u002Fwatch?v=Lv1iVwlnqTk","wrKUFmk_NrDjD2XFWGTHv2IdVTP9zppvFFJEEw1QhgA",[89],{"id":90,"title":91,"date":92,"draft":17,"englishVersion":16,"extension":93,"language":22,"meta":94,"originalTitle":95,"outlet":96,"stem":97,"summary":98,"url":99,"__hash__":100},"writing\u002Fwriting\u002Fwhip-standard-protocol.yml","WHIP: a standard signalling protocol for WebRTC","2025-07-23","yml",{},"WHIP — стандартный протокол общения WebRTC приложений","Habr, Yandex Cloud & Infrastructure blog","writing\u002Fwhip-standard-protocol","Why WebRTC ingest needs a standard signalling protocol and how WHIP works: signalling versus media, the HTTP exchange (OPTIONS, POST, PATCH, DELETE), ETag and If-Match, trickle ICE, and the current limits around BUNDLE, Plan B, DTLS roles and renegotiation.","https:\u002F\u002Fhabr.com\u002Fru\u002Fcompanies\u002Fyandex_cloud_and_infra\u002Farticles\u002F929936\u002F","wYxduJuurDnJDvDA1N04m_FZVqPbqEdG300y6-RgVxk",[102,119,137],{"id":103,"draft":17,"extension":93,"highlights":104,"links":107,"meta":109,"name":110,"order":111,"status":16,"stem":112,"tagline":113,"tags":114,"__hash__":118},"projects\u002Fprojects\u002Fwebrtc-page.yml",[105,106],"11 codebases: libwebrtc across Chrome milestones, Pion, coturn, Kamailio, rtpengine, Asterisk, FreeSWITCH, sipXcom, libdatachannel, Janus, Jitsi","Semantic code search, AST symbol navigation, version diffs, build-graph inspection, RFC lookup, SDP\u002FSIP parsing",{"site":108},"https:\u002F\u002Fwebrtc.page\u002F",{},"webrtc.page","1","projects\u002Fwebrtc-page","MCP servers that give AI assistants structured access to WebRTC and VoIP source code, version by version.",[115,116,117],"MCP","libwebrtc","Pion","JXsGx2WhSn61W2EjO7zOxmfljWHwjh1X8_TCwqIMrEI",{"id":120,"draft":17,"extension":93,"highlights":121,"links":125,"meta":128,"name":129,"order":130,"status":16,"stem":131,"tagline":132,"tags":133,"__hash__":136},"projects\u002Fprojects\u002Fsimple-sdp.yml",[122,123,124],"Typed ICE, DTLS, BUNDLE, simulcast and codecs","Tested on real Chrome, Firefox and SIP-device SDP","TypeScript with zero dependencies; WebAssembly, Swift and Kotlin builds",{"github":126,"npm":127},"https:\u002F\u002Fgithub.com\u002Firbisadm\u002Fsimple-sdp","https:\u002F\u002Fwww.npmjs.com\u002Fpackage\u002F@irbisadm\u002Fsimple-sdp",{},"simple-sdp","3","projects\u002Fsimple-sdp","Strict, lossless SDP parser, writer and validator (RFC 8866) for WebRTC and SIP.",[134,135],"SDP","TypeScript","7wf5nad_YqL_3h8De3clrSibaYUbiW6xwmUgXCK3Rm8",{"id":138,"draft":17,"extension":93,"highlights":139,"links":141,"meta":143,"name":144,"order":145,"status":16,"stem":146,"tagline":147,"tags":148,"__hash__":151},"projects\u002Fprojects\u002Frtcstats-proto.yml",[140],"A compact typed wire format for shipping getStats() snapshots to a collector",{"github":142},"https:\u002F\u002Fgithub.com\u002Firbisadm\u002FRTCStats.proto",{},"RTCStats.proto","4","projects\u002Frtcstats-proto","Protocol Buffers schema for the W3C WebRTC Statistics API.",[149,150],"Protobuf","getStats","NCizozyNlct4heIvZtITxkUeNpgET2QMrBztmcp2maA",1794079529332]